component related function should be implemented at
soc-component.c.
This patch adds snd_soc_component_compr_get_codec_caps().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/877dqp7v6i.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
component related function should be implemented at
soc-component.c.
This patch adds snd_soc_component_compr_get_caps().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/878sb57v6q.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
component related function should be implemented at
soc-component.c.
This patch adds snd_soc_component_compr_get_params().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/87a6vl7v6x.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
component related function should be implemented at
soc-component.c.
This patch moves soc-compress soc_compr_components_set_params()
to soc-component as snd_soc_component_compr_set_params().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/87blg17v74.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
component related function should be implemented at
soc-component.c.
This patch moves soc-compress soc_compr_components_trigger()
to soc-component as snd_soc_component_compr_trigger().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/87d00h7v7k.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
component related function should be implemented at
soc-component.c.
This patch moves soc-compress soc_compr_components_free()
to soc-component as snd_soc_component_compr_free().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/87eekx7v7r.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
component related function should be implemented at
soc-component.c.
This patch moves soc-compress soc_compr_components_open()
to soc-component as snd_soc_component_compr_open().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/87ft5d7v7x.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In kabylake_set_bias_level(), enabling mclk may fail if the clock has
already been enabled by the firmware. Attempts to disable that clock
later will fail with a warning backtrace.
mclk already disabled
WARNING: CPU: 2 PID: 108 at drivers/clk/clk.c:952 clk_core_disable+0x1b6/0x1cf
...
Call Trace:
clk_disable+0x2d/0x3a
kabylake_set_bias_level+0x72/0xfd [snd_soc_kbl_rt5663_rt5514_max98927]
snd_soc_card_set_bias_level+0x2b/0x6f
snd_soc_dapm_set_bias_level+0xe1/0x209
dapm_pre_sequence_async+0x63/0x96
async_run_entry_fn+0x3d/0xd1
process_one_work+0x2a9/0x526
...
Only disable the clock if it has been enabled.
Fixes: 15747a8020 ("ASoC: eve: implement set_bias_level function for rt5514")
Cc: Brent Lu <brent.lu@intel.com>
Cc: Curtis Malainey <cujomalainey@chromium.org>
Signed-off-by: Guenter Roeck <linux@roeck-us.net>
Link: https://lore.kernel.org/r/20201111205434.207610-1-linux@roeck-us.net
Signed-off-by: Mark Brown <broonie@kernel.org>
The Intel Keem Bay audio module is only present on Intel Keem Bay SoCs.
Hence add a dependency on ARCH_KEEMBAY, to prevent asking the user about
this driver when configuring a kernel without Intel Keem Bay platform
support.
Fixes: c544912bcc ("ASoC: Intel: Add makefiles and kconfig changes for KeemBay")
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://lore.kernel.org/r/20201110145001.3280479-1-geert+renesas@glider.be
Signed-off-by: Mark Brown <broonie@kernel.org>
The Freescale/NXP AUDIO TO HDMI TX module is only present on NXP i.MX 8
Series SoCs. Hence add a dependency on ARCH_MXC, to prevent asking the
user about this driver when configuring a kernel without i.MX 8 platform
support.
Fixes: 8a24c834c0 ("ASoC: fsl_aud2htx: Add aud2htx module driver")
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://lore.kernel.org/r/20201110145120.3280658-1-geert+renesas@glider.be
Signed-off-by: Mark Brown <broonie@kernel.org>
Audio ADI reference designs are also used on some ZynqMP boards, and can
also be used on Intel FPGA boards and also on some more complex FPGA
combinations (FPGA cards connected through PCIe).
This change removes the dependency on Microblaze and Zynq architectures
to allow the usage of this driver for the systems described above.
Signed-off-by: Bogdan Togorean <bogdan.togorean@analog.com>
Signed-off-by: Alexandru Ardelean <alexandru.ardelean@analog.com>
Link: https://lore.kernel.org/r/20201110152213.37811-1-alexandru.ardelean@analog.com
Signed-off-by: Mark Brown <broonie@kernel.org>
First two of the series address bugs connected mainly to offload streams:
- scenarios with very low buffer sizes: RESET_STREAM IPC timeouts
- fix lp clock selection when switching between PAUSE <-> RESUME states:
glitches on first offload when no additional stream is opened
simultaneously
Follow ups are: code reduction and optimization oriented patches.
This has been foretold in:
[PATCH v10 00/14] ASoC: Intel: Catpt - Lynx and Wildcat point
https://www.spinics.net/lists/alsa-devel/msg116440.html
Note: LPT power up/down sequences might get aligned with WPT once
enough testing is done as capabilities are shared for both DSPs.
First, optimize applying of user settings - prevent redundand calls from
happening - and then as mentioned above, streamline power on/off sequence
for LPT and WPT.
Cezary Rojewski (5):
ASoC: Intel: catpt: Skip position update for unprepared streams
ASoC: Intel: catpt: Correct clock selection for dai trigger
ASoC: Intel: catpt: Optimize applying user settings
ASoC: Intel: catpt: Streamline power routines across LPT and WPT
ASoC: Intel: catpt: Cleanup after power routines streamlining
sound/soc/intel/catpt/core.h | 10 ++-
sound/soc/intel/catpt/device.c | 18 +++---
sound/soc/intel/catpt/dsp.c | 56 ++--------------
sound/soc/intel/catpt/pcm.c | 113 ++++++++++++++++-----------------
4 files changed, 74 insertions(+), 123 deletions(-)
--
2.17.1
base-commit: 3650b228f8
Set format from `set_fmt()` func instead of `hw_params()`, plus supportive
commits
Kirill Marinushkin (4):
ASoC: pcm512x: Fix not setting word length if DAIFMT_CBS_CFS
ASoC: pcm512x: Rearrange operations in `hw_params()`
ASoC: pcm512x: Move format check into `set_fmt()`
ASoC: pcm512x: Add support for more data formats
sound/soc/codecs/pcm512x.c | 134 ++++++++++++++++++++++++++++-----------------
1 file changed, 84 insertions(+), 50 deletions(-)
--
2.13.6
Add 32 bit format support for capture in lpass-sc7180
snd_soc_dai_driver capabilities. Need to add contstraints
in machine driver so that only specific format allowed.
Signed-off-by: V Sujith Kumar Reddy <vsujithk@codeaurora.org>
Signed-off-by: Srinivasa Rao Manidadapu <srivasam@codeaurora.org>
Link: https://lore.kernel.org/r/1605512955-7017-1-git-send-email-srivasam@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Kconfig entries, PCI ID and chip info for Alderlake-S product.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20201116142642.2106067-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When compiled with CONFIG_HAVE_CLK, the kernel need to get provider for the
clock API. This is usually selected by the platform and the sound drivers
should not really care about this. However COMPILE_TEST is special and the
platform required may not have been selected, leading to this type of
error:
> aiu-encoder-spdif.c:(.text+0x3a0): undefined reference to `clk_set_parent'
Since we need a sane provider of the API with COMPILE_TEST, depends on
COMMON_CLK.
Fixes: 6dc4fa179f ("ASoC: meson: add axg fifo base driver")
Reported-by: kernel test robot <lkp@intel.com>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20201116172423.546855-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The "no_headphone" variable is never set to "false" so it could be
uninitialized.
Fixes: e936619b7c ("ASoC: qcom: sc7180: Modify machine driver for sound card")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20201116125950.GA44063@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
With LPT switching to WPT-based power on/off routines, functions that
have been previously used by it are rendered redundant so remove them.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20201116133332.8530-6-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no need for separate power on/off routines for LPT and WPT as
as the protocol is shared for both platforms. Make WPT routines generic
and reuse them in LPT case too.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20201116133332.8530-5-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Initial user settings such as volume control need to be applied only
once after stream is allocated. As prepare() operation can be invoked
multiple times during the stream's lifetime, relocate
catpt_dai_apply_usettings() and call it directly within
catpt_dai_hw_params() rather than on every catpt_dai_prepare().
catpt_dai_apply_usettings() remains unchanged.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20201116133332.8530-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
During stream start DSP firmware requires LPCS disabled as that moment in
time is resource heavy. Currently high-clock is selected on start of
second stream onwards while low-clock is re-selected before stream
actually leaves RESUME state i.e. PAUSE_STREAM call. Fix this by always
updating clock before RESUME_STREAM and directly after PAUSE_STREAM.
Fixes: a126750fc8 ("ASoC: Intel: catpt: PCM operations")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20201116133332.8530-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Playing with very low period sizes may lead to timeouts when awaiting
RESET_STREAM reply for offload streams. This is caused by NOTIFY_POSITION
appearing in the middle of trigger(stop).
Stream is unprepared during trigger(stop) where PAUSE_STREAM IPC gets
invoked. However, all data that is already mixed in DSP firmware's mixer
stream will still be played regardless of the pause. For offload streams,
this means possibility for another NOTIFY_POSITION to process. Keep these
notifications in check by only handling them when stream is in prepared
state.
Fixes: a126750fc8 ("ASoC: Intel: catpt: PCM operations")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20201116133332.8530-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, pcm512x driver supports only I2S data format.
This commit adds RJ, LJ, DSP_A and DSP_B as well.
I don't expect regression WRT existing sound cards, because:
* default value in corresponding register of pcm512x codec is 0 == I2S
* existing in-tree sound cards with pcm512x codec are configured for I2S
* i don't see how existing off-tree sound cards with pcm512x codec could be
configured differently - it would not work
* tested explicitly, that there is no regression with Raspberry Pi +
sound card `sound/soc/bcm/hifiberry_dacplus.c`
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Matthias Reichl <hias@horus.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20201115122306.18164-5-kmarinushkin@birdec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
I would like to describe the reasoning by quoting Peter Ujfalusi
<peter.ujfalusi@ti.com> from his review of this patch series v1 [1]:
> When you bind a link you will use set_fmt for the two sides to see if
> they can agree, that both can support what has been asked.
>
> The pcm512x driver just saves the fmt and say back to that card:
> whatever, I'm fine with it. But runtime during hw_params it can fail due
> to unsupported bus format, which it actually acked to be ok.
>
> This is the difference.
>
> Sure, some device have constraint based on the fmt towards the hw_params
> and it is perfectly OK to do such a checks and rejections or build
> rules/constraints based on fmt, but failing hw_params just because
> set_fmt did not checked that the bus format is not even supported is not
> a nice thing to do.
[1] https://patchwork.kernel.org/project/alsa-devel/patch/
20201109212133.25869-1-kmarinushkin@birdec.com/
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Matthias Reichl <hias@horus.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20201115122306.18164-4-kmarinushkin@birdec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This commit is a preparation for the next patch in the series.
It's goal is to make format check easy-to-move-out. Theoretically, more
butifications are possile in `hw_params()` func, but my intention in this
commit is to keep behaviour unchanged.
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Matthias Reichl <hias@horus.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20201115122306.18164-3-kmarinushkin@birdec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In `pcm512x_hw_params()`, the following switch-case:
~~~~
switch (pcm512x->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
~~~~
returns 0, which was preventing word length from being written into codecs
register.
Fixed by writing it into register before checking
`SND_SOC_DAIFMT_MASTER_MASK`.
Tested with Raspberry Pi + sound card `hifiberry_dacplus` in CBS_CFS format
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Matthias Reichl <hias@horus.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20201115122306.18164-2-kmarinushkin@birdec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
lpass_pcm_data is not freed in error paths. Free it in
error paths to avoid memory leak.
Fixes: 022d00ee0b ("ASoC: lpass-platform: Fix broken pcm data usage")
Signed-off-by: Pavel Machek <pavel@ucw.cz>
Signed-off-by: V Sujith Kumar Reddy <vsujithk@codeaurora.org>
Signed-off-by: Srinivasa Rao Mandadapu <srivasam@codeaurora.org>
Link: https://lore.kernel.org/r/1605416210-14530-1-git-send-email-srivasam@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
S24_LE is 24 bit audio in 32 bit container configuration
Fixing the configuration to match the data arrangement of
this audio format.
Fixes: c5477e9667 ("ASoC: Intel: Add KeemBay platform driver")
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20201116061905.32431-2-michael.wei.hong.sit@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The SOF (Sound Open Firmware) tree contains a lot of references in
topology files to 'codec_slave'/'codec_master' terms, which in turn
come from alsa-lib and ALSA/ASoC topology support at the kernel
level. These terms are no longer compatible with the guidelines
adopted by the kernel community [1], standard organizations, and need
to change in backwards-compatible ways.
The main/secondary terms typically suggested in guidelines don't mean
anything for clocks, this patchset suggests instead the use of
'provider' and 'consumer' terms, with the 'codec' prefix kept to make
it clear that the codec is the reference. The CM/CS suffixes are also
replaced by CP/CC.
It can be argued that the change of suffix is invasive, but finding a
replacement that keeps the M and S shortcuts has proven difficult in
quite a few contexts.
The previous definitions are kept for backwards-compatibility so this
change should not have any functional impact. It is suggested that new
contributions only use the new terms but there is no requirement to
transition immediately to the new definitions for existing code. Intel
will however update all its past contributions related to bit
clock/frame sync configurations immediately.
This patchset contains the kernel changes only, the alsa-lib changes
were shared separately.
Feedback welcome
~Pierre
[1] https://lkml.org/lkml/2020/7/4/229
Pierre-Louis Bossart (4):
ASoC: topology: use inclusive language for bclk and fsync
ASoC: SOF: use inclusive language for bclk and fsync
ASoC: Intel: atom: use inclusive language for SSP bclk/fsync
ASoC: Intel: keembay: use inclusive language for bclk and fsync
include/sound/soc-dai.h | 32 +++++++++++++++---------
include/sound/sof/dai.h | 16 ++++++++----
include/uapi/sound/asoc.h | 22 ++++++++++------
sound/soc/intel/atom/sst-atom-controls.c | 12 ++++-----
sound/soc/intel/atom/sst-atom-controls.h | 4 +--
sound/soc/intel/keembay/kmb_platform.c | 22 ++++++++--------
sound/soc/intel/keembay/kmb_platform.h | 8 +++---
sound/soc/soc-topology.c | 24 +++++++++---------
sound/soc/sof/topology.c | 18 ++++++-------
9 files changed, 89 insertions(+), 69 deletions(-)
--
2.25.1
To meet the most platform, the detection time should be increased to
avoid that the flushing DAC data fails.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20201113092125.19206-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The imx-ssi driver was only used by i.MX non-DT platforms.
Since 5.10-rc1, i.MX has been converted to a DT-only platform and all
board files are gone.
Remove the imx-ssi audio driver as there are no more users at all.
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20201110203937.25684-6-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SOC_IMX_SSI was only used by i.MX non-DT platforms.
SND_SOC_EUKREA_TLV320 already selects the SND_SOC_FSL_SSI symbol,
which is enough.
Remove the unneeded SND_SOC_IMX_SSI selection.
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20201110203937.25684-5-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The imx-mc13783 was used on imx27-pdk and imx31-pdk non-DT platforms.
Since 5.10-rc1, i.MX has been converted to a DT-only platform and all
board files are gone.
Remove the imx-mc13783 audio machine driver as there is no user at all.
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20201110203937.25684-4-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Since commit e1324ece2a ("ARM: imx: Remove i.MX35 board files"), the
MACH_PCM043 and MACH_PCA100 non-DT platform are no longer supported,
so get rid of their machine audio driver too.
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20201110203937.25684-3-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Since commit 879c0e5e0a ("ARM: imx: Remove i.MX27 board files"), the
MACH_IMX27_VISSTRIM_M10 non-DT platform is no longer supported,
so get rid of its machine audio driver too.
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20201110203937.25684-2-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Since commit c93197b004 ("ARM: imx: Remove i.MX31 board files"), the
MACH_MX31ADS_WM1133_EV1 non-DT platform is no longer supported,
so get rid of its machine audio driver too.
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20201110203937.25684-1-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_INTEL_DSP_CONFIG is selected by the HDaudio, Skylake and SOF
drivers. When the HDaudio link is not selected as a option, this
Kconfig option is not touched and will default to whatever other
drivers selected. In the case e.g. where HDaudio is compiled as
built-in, the linker will complain:
ld: sound/soc/sof/sof-pci-dev.o: in function `sof_pci_probe':
sof-pci-dev.c:(.text+0x5c): undefined reference to
`snd_intel_dsp_driver_probe'
Adding the select for all HDaudio platforms, regardless of whether
they rely on the HDaudio link or not, solves the problem.
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Acked-by: Randy Dunlap <rdunlap@infradead.org>
Fixes: 82d9d54a6c ('ALSA: hda: add Intel DSP configuration / probe code')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20201112164425.25603-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add periods at the end of sentences in help text.
Insert a "not" in one place.
Fix help text indentation.
Remove a duplicate "Intel(R)" in one place.
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20201112164425.25603-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add periods at the end of sentences in help text.
Drop "a" in one place where it is not needed.
Fix a verb.
Add some capitalization.
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20201112164425.25603-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add periods at the end of sentences in help text.
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20201112164425.25603-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The firmware extended data IPC and manifest structures are designed to
be extendable without breaking the driver-firmware ABI. Driver should
not raise a warning in case a new header type is detected at
firmware boot. There are already checks for IPC ABI compatibility in
snd_sof_ipc_valid() and if the versions are deemed compatible, extra
fields in IPC messages should not trigger warnings.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Karol Trzciński <karolx.trzcinski@linux.intel.com>
Link: https://lore.kernel.org/r/20201111173321.1933452-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Smart_amp_speaker device has the playback stream and capture stream
associated to it. Hence add the dpcm_capture = 1 flag while dailink
creation.
This patches fixes:
ERR kernel [timestamp] SSP1-Codec: ASoC: no backend capture stream
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Dharageswari R <dharageswari.r@intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20201109210958.84198-1-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It's not clear why this driver has an additional call to
regmap_mark_dirty(), remove to align with others.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20201111214318.150529-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
cppcheck complains, use separate loop variable for sink and source ports
sound/soc/codecs/rt711-sdw.c:382:4: style: Variable 'i' is reassigned
a value before the old one has been used. [redundantAssignment]
i = 0;
^
sound/soc/codecs/rt711-sdw.c:371:4: note: i is assigned
i++;
^
sound/soc/codecs/rt711-sdw.c:382:4: note: i is overwritten
i = 0;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Link: https://lore.kernel.org/r/20201111214318.150529-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Pointer cast is not necessary, remove.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Link: https://lore.kernel.org/r/20201111214318.150529-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In the normal path, we should not free the arizona,
we should return immediately. It will be free when
call remove operation.
Fixes: 31833ead95 ("ASoC: arizona: Move request of speaker IRQs into bus probe")
Reported-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Zhang Qilong <zhangqilong3@huawei.com>
Acked-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20201111130923.220186-2-zhangqilong3@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The pm_runtime_enable will increase power disable depth. Thus
a pairing decrement is needed on the error handling path to
keep it balanced according to context. This series of patches
fixed it.
Zhang Qilong (3):
ASoC: wm8994: Fix PM disable depth imbalance on error
ASoC: wm8997: Fix PM disable depth imbalance on error
ASoC: wm8998: Fix PM disable depth imbalance on error
sound/soc/codecs/wm8994.c | 6 +++++-
sound/soc/codecs/wm8997.c | 9 +++++++--
sound/soc/codecs/wm8998.c | 4 +++-
3 files changed, 15 insertions(+), 4 deletions(-)
--
2.25.4
We can choose to drop away any length of data from the beginning according
to project needs. Some projects don't want to throw away any data, because
they want to use recorded data to do echo cancellation, so they have to
make sure that they are aligned with the reference data as much as
possible. Or there are other algorithms in the upper layer to eliminate
this noise. Or some projects want to eliminate this noise form the kernel
layer. However, the minimum recommended value is 50ms to skip pop noise.
Signed-off-by: Jiaxin Yu <jiaxin.yu@mediatek.com>
Link: https://lore.kernel.org/r/1605062722-8711-1-git-send-email-jiaxin.yu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Deferred probe is an expected return value for snd_soc_register_card().
Given that the driver deals with it properly, there's no need to output
a warning that may potentially confuse users.
Signed-off-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/20201111103245.152189-1-jonathanh@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The pm_runtime_enable will increase power disable depth. Thus
a pairing decrement is needed on the error handling path to
keep it balanced according to context.
Fixes: 31833ead95 ("ASoC: arizona: Move request of speaker IRQs into bus probe")
Signed-off-by: Zhang Qilong <zhangqilong3@huawei.com>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20201111041326.1257558-4-zhangqilong3@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The pm_runtime_enable will increase power disable depth. Thus
a pairing decrement is needed on the error handling path to
keep it balanced according to context.
Fixes: 57e265c8d7 ("ASoC: wm8994: Move runtime PM init to platform device init")
Signed-off-by: Zhang Qilong <zhangqilong3@huawei.com>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20201111041326.1257558-2-zhangqilong3@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This series is a prepraration for using generic graph driver for Tegra210
audio. Tegra audio graph series will be sent in a separate series because
it has some dependency over other series for documentation work. This
series can focus on the generic ASoC driver updates. Below are the summary
of changes done.
* Support multiple instances of a component. For example there can be
multiple I2S devices which can use the same component driver.
* Support open platforms with empty Codec endpoint. Customers can plug
their own HW and can populate codec endpoint.
* In a component model there can be many components which can be
connected together. In such cases Identify no-pcm DPCM DAI links which
can be used in BE<->BE connections.
* Expose structures or helpers to be re-used by other similar graph
drivers.
The series is based on following references where DPCM usgae for Tegra
Audio and simple-card driver proposal were discussed.
* https://lkml.org/lkml/2020/4/30/519 (DPCM for Tegra)
* https://lkml.org/lkml/2020/6/27/4 (simple-card driver)
Changelog
=========
v4 -> v5
--------
* No changes in the core/audio-graph driver patches which are
part of the current series.
* Dropped Tegra audio patches and doc patches. This will be
sent separately once the doc depdendencies are resolved.
v3 -> v4
--------
* Added new patches to convert graph.txt and audio-graph-card.txt
to corresponding json-schema files. Later these references
are used in Tegra audio graph schema.
* AHUB component binding docs are updated to reflect the usage
of ports/port/endpoint
* More common stuff is moved into graph_parse_of() and this is
used by both generic and Tegra audio graph.
* DT binding for Tegra audio graph is updated to included "ports { }"
* As per the suggestion 'void *data' member is dropped from
'asoc_simple_priv' and instead container method is used to
maintain required custom data internal to Tegra audio graph.
v2 -> v3
--------
* Dropped new compatible addition in generic graph driver
after reviewing it with Morimoto-san. Instead added Tegra
audio graph driver and new compatibles are added in the same.
* Added new patches to expose new members for customization
in audio graph driver.
* Added new patch for Tegra audio graph driver and related
documentation.
* Minor change in below commit where mutex version of helper is used
"ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM"
* DT binding is updated to use the newly exposed compatibles
* No changes in other patches
v1 -> v2
--------
* Re-organized ports/endpoints description for ADMAIF and XBAR.
Updated DT patches accordingly.
* After above change, multiple Codec endpoint support is not
required and hence dropped for now. This will be considered
separately if at all required in future.
* Re-ordered patches in the series.
Sameer Pujar (7):
ASoC: soc-core: Fix component name_prefix parsing
ASoC: soc-pcm: Get all BEs along DAPM path
ASoC: audio-graph: Use of_node and DAI for DPCM DAI link names
ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM
ASoC: audio-graph: Support empty Codec endpoint
ASoC: audio-graph: Expose new members for asoc_simple_priv
ASoC: audio-graph: Expose helpers from audio graph
include/sound/graph_card.h | 16 ++++
include/sound/simple_card_utils.h | 3 +
include/sound/soc.h | 1 +
sound/soc/generic/audio-graph-card.c | 175 ++++++++++++++++++++++++-----------
sound/soc/soc-core.c | 3 +-
sound/soc/soc-pcm.c | 3 +-
6 files changed, 143 insertions(+), 58 deletions(-)
create mode 100644 include/sound/graph_card.h
--
2.7.4
There are two issues in this function.
1) We can't drop the refrences on "cpu", "codec" and "platform" before
we take the reference. This doesn't cause a problem on the first
iteration because those pointers start as NULL so the of_node_put()
is a no-op. But on the subsequent iterations, it will lead to a use
after free.
2) If the devm_kzalloc() allocation failed then the code returned
directly instead of cleaning up.
Fixes: c1e6414cdc ("ASoC: qcom: common: Fix refcount imbalance on error")
Fixes: 1e36ea360a ("ASoC: qcom: common: use modern dai_link style")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20201105125154.GA176426@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
The break condition copied by mistake as same
as loop condition in the previous version, but must
be the opposite. So fix it.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Link: https://lore.kernel.org/r/20201102161810.902464-1-viorel.suman@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This commit exposes following functions which can be used by a sound
card driver based on generic audio graph driver. Idea is vendors can
have a thin driver and re-use common stuff from audio graph driver.
- graph_card_probe()
- graph_parse_of()
In doing so a new header file is added for above. The graph_probe()
function is simplified by moving more common stuff to graph_parse_of().
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/1604329814-24779-8-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add new members in struct 'asoc_simple_priv'. Idea is to leverage
simple or graph card driver as much as possible and vendor can
maintain a thin driver to control the behavior by populating these
newly exposed members.
Following are the members added in 'asoc_simple_priv':
- 'ops' struct: In some cases SoC vendor drivers may want to
implement 'snd_soc_ops' callbacks differently. In such cases
custom callbacks would be used.
- 'force_dpcm' flag: Right now simple or graph card drivers
detect DAI links as DPCM links if:
* The dpcm_selectable is set AND
* Codec is connected to multiple CPU endpoints or aconvert
property is used for rate/channels.
So there is no way to directly specify usage of DPCM alone. So a
flag is exposed to mark all links as DPCM. Vendor driver can
set this if required.
- 'dpcm_selectable': Currently simple or audio graph drivers
provide a way to enable this for specific compatibles. However
vendor driver may want to define some additional info. Thus
expose this variable where vendor drivers can set this if
required.
Audio graph driver is updated to consider above flags or callbacks.
Subsequent patches in the series illustrate usage for above.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/1604329814-24779-7-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For open platforms, which can support pluggable audio cards, Codec
endpoint is not fixed always. It actually depends on the compatible
HW module that is going to be connected. From SoC side the given I/O
interface is always available. Hence such links have fixed CPU endpoint
but no Codec endpoint. This patch helps to support such links where
user can populate Codec endpoint only and its fields in Platform DT
depending on the plugged HW.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/1604329814-24779-6-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
PCM devices are created for FE dai links with 'no-pcm' flag as '0'.
Such DAI links have CPU component which implement either pcm_construct()
or pcm_new() at component or dai level respectively. Based on this,
current patch exposes a helper function to identify such components
and populate 'no_pcm' flag for DPCM DAI link.
This helps to have BE<->BE component links where PCM devices need
not be created for CPU component involved in such links.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/1604329814-24779-5-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For multiple instances of components, using DAI name alone for DAI links
is causing conflicts. Components can define multiple DAIs and hence using
just a device name won't help either. Thus DT device node reference and
DAI names are used to uniquely represent DAI link names.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/1604329814-24779-4-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
dpcm_end_walk_at_be() stops the graph walk when first BE is found for
the given FE component. In a component model we may want to connect
multiple DAIs from different components. A new flag is introduced in
'snd_soc_card', which when set allows DAI/component chaining. Later
PCM operations can be called for all these listed components for a
valid DAPM path.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1604329814-24779-3-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The "prefix" can be defined in DAI link node or it can be specified as
part of the component node itself. Currently "sound-name-prefix" defined
in a component is not taking effect. Actually the property is not getting
parsed. It can be fixed by parsing "sound-name-prefix" property whenever
"prefix" is missing in DAI link Codec node.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1604329814-24779-2-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Note:
- The patch is made by the collaboration of
Ajye Huang <ajye_huang@compal.corp-partner.google.com>
Cheng-Yi Chiang <cychiang@chromium.org>
v6:
- Documentation: Addressed suggestions from Rob Herring.
- Define "maxItems: 1" in dmic-gpios property.
- Only keep one example and add dmic-gpios property in.
v5:
- Machine driver:
- Fix a format string warning (Reported-by: kernel test robot <lkp@intel.com>).
detailed info at https://lore.kernel.org/patchwork/patch/1331087/
v4:
- Machine driver: Addressed suggestions from Tzung-Bi.
- Remove redundant judgments in dmic_set() and dmic_get().
- Remove 1 level indent of judgment of IS_ERR(data->dmic_sel).
v3:
- Machine driver: Addressed suggestions from Tzung-Bi.
- Move variables "dmic_switch" and "dmic_sel" into struct sc7180_snd_data.
- Remove redundant judgments in dmic_set().
v2:
- Documentation: Modify the dimc-gpios property description and examples.
- Machine driver:
- Remove "qcom,sc7180-sndcard-rt5682-m98357-2mic" compatible
- See gpio property and use anadditional control.
Thanks for the review!
Ajye Huang (2):
ASoC: google: dt-bindings: modify machine bindings for two MICs case
ASoC: qcom: sc7180: Modify machine driver for 2mic
.../bindings/sound/google,sc7180-trogdor.yaml | 8 ++-
sound/soc/qcom/sc7180.c | 61 +++++++++++++++++++
2 files changed, 68 insertions(+), 1 deletion(-)
--
2.25.1
"Shane.Chien" <shane.chien@mediatek.com>:
From: "Shane.Chien" <shane.chien@mediatek.com>
This series of patches is to fix vaud18 power leakage problem.
vaud18 will be enable only when mt6359 audio path is turned on.
Change since v2:
- fix dt-binnding syntex error
Change since v1:
- use dapm regulator supply widget for vaud18 control.
- add vaud18 regulator property in mt6359 dt-binding.
Shane.Chien (2):
ASoC: Fix vaud18 power leakage of mt6359
dt-bindings: mediatek: mt6359: Add new property for mt6359
.../devicetree/bindings/sound/mt6359.yaml | 9 +++++++
sound/soc/codecs/mt6359.c | 25 +-------------------
sound/soc/codecs/mt6359.h | 8 -------
3 files changed, 10 insertions(+), 32 deletions(-)
--
1.7.9.5
Hi Mark
These are v2 of tidyup patch-set for SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx.
soc-compress is using both SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx, but mixed use.
This is confusable, but no problem. Because these are defined as UAPI and
are using same value.
This patch-set make sure these are same value.
v1 -> v2
- checks COMPRESS vs PCM_STREAM by using BUILD_BUG_ON()
Link: https://lore.kernel.org/r/87wnzcfnkk.wl-kuninori.morimoto.gx@renesas.com
Kuninori Morimoto (2):
ASoC: soc-compress: tidyup STREAM vs COMPRESS
ASoC: soc-compress: assume SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx are same
sound/soc/soc-compress.c | 63 +++++++++++++++-------------------------
1 file changed, 23 insertions(+), 40 deletions(-)
--
2.25.1
Add codec driver for Nuvoton NAU8315.
Signed-off-by: David Lin <CTLIN0@nuvoton.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20201102023212.594137-1-CTLIN0@nuvoton.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc-compress.c is using both SND_COMPRESS_xxx and SND_PCM_STREAM_xxx.
These are defined as UAPI, and has same value.
enum {
SNDRV_PCM_STREAM_PLAYBACK = 0,
SNDRV_PCM_STREAM_CAPTURE,
...
};
enum snd_compr_direction {
SND_COMPRESS_PLAYBACK = 0,
SND_COMPRESS_CAPTURE
};
Essentially both COMPRESS and PCM_STREAM definitions are identical,
and can be never different because of ABI compatibility,
which means it's safe to mix both variants in the code.
This patch checks it by BUILD_BUG_ON(), and merge to use.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87361wmsyg.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_runtime_activate() and
snd_soc_dai_digital_mute() need SNDRV_PCM_STREAM_xxx
instead of SND_COMPRESS_xxx.
These are bug but nothing happen because these are same value.
enum {
SNDRV_PCM_STREAM_PLAYBACK = 0,
SNDRV_PCM_STREAM_CAPTURE,
...
};
enum snd_compr_direction {
SND_COMPRESS_PLAYBACK = 0,
SND_COMPRESS_CAPTURE
};
This patch tidyup it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/874kmcmsyn.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In addition, having mixer control to switch between DMICs by
using "dmic-gpios" property.
Refer to this one as an example,
commit b7a742cff3 ("ASoC: AMD: Use mixer control to switch between DMICs")
Signed-off-by: Ajye Huang <ajye_huang@compal.corp-partner.google.com>
Reviewed-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20201106061433.1483129-3-ajye_huang@compal.corp-partner.google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
vaud18 is power of mt6359 audio path. It
should only enable when audio is used,
instead of in boot up stage.
Once mt6359 audio path is enabled or disabled,
vaud18 is controlled by regulator supply widget
"LDO_VAUD18". Due to vaud18 is controlled by
regulator dapm macro instead of regmap, the macro
MT6359_LDO_VAUD18_CON0 and variable avdd_reg
is no used and removed from mt6359.h.
Signed-off-by: Shane.Chien <shane.chien@mediatek.com>
Link: https://lore.kernel.org/r/1604975492-6142-2-git-send-email-shane.chien@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Hi,
The series is inspired by the effort to standardize TI's arm64 dtsi files to keep
all nodes in 'okay' state and let the board dts files disable not needed
peripherals (and not properly configured):
https://lore.kernel.org/lkml/20201104224356.18040-1-nm@ti.com/
In the unlikely (or likely?) event when the dts misses to disable the McASP node
which is not configured we currenly and luckily just manage to not crash as we
had fixup code in place for legacy pdata boots.
This however prints out a message which does not really help to identify the
issue.
This series will reduce some of the noise during boot (first patch) then
adds the needed changes to handle the variations of 'okay':
A - have all required DT properties for audio
B - gpiochip is enabled
A && !B -> everything is OK for audio, no gpiochip registered
A && B -> everything is OK for audio, gpiochip is registered
!A && B -> audio is not OK, gpiochip is registered. dev_dbg() for audio and do
not register SOC DAI and PCM
!A && !B -> audio is not OK, no gpiochip. dev_err() and fail the probe
Regards,
Peter
---
Peter Ujfalusi (4):
ASoC: ti: davinci-mcasp: Use platform_get_irq_byname_optional
ASoC: ti: davinci-mcasp: Remove legacy dma_request parsing
ASoC: ti: davinci-mcasp: Simplify the configuration parameter handling
ASoC: ti: davinci-mcasp: Handle missing required DT properties
sound/soc/ti/davinci-mcasp.c | 294 ++++++++++++++---------------------
1 file changed, 119 insertions(+), 175 deletions(-)
--
Peter
Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki.
Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki
platform_get_irq() would print error message internally, so dev_err()
after platform_get_irq() is not needed
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1604715643-29507-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
McASP needs three required properties to be usable for audio:
op-mode, tdm-slots and the serial-dir array.
Instead of probing the driver even without the needed information we should
make sure that all the parameters are provided for operation.
The fact that McASP can act as a GPIO controller for it's pins complicates
this a bit, but as a general rule we can:
- we fail the probe if McASP is not configured to be used as gpiochip
- we will not register the DAI (and PCM) if gpiochip is defined
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20201106072551.689-5-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace the davinci_mcasp_set_pdata_from_of() function which returned a
pdata pointer with davinci_mcasp_get_config() to return an actual error
code and handle all pdata validation and private mcasp struct setup in
there.
Drop the unused ram-size-playback and sram-size-capture query from DT at
the same time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20201106072551.689-4-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The legacy dma_request (which was holding the DMA request number) is no
longer in use for a long time.
All legacy platforms has been converted to dma_slave_map.
Remove it along with the DT parsing to get tx_dma_channel and
rx_dma_channel.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20201106072551.689-3-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Depending on the integration of McASP either the 'common' or the
'rx' and 'tx' or only the 'tx' interrupt number is valid, provided.
By switching to platform_get_irq_byname_optional() we can clean up the
bootlog from messages like:
davinci-mcasp 2ba0000.mcasp: IRQ common not found
The irq number == 0 is not valid, fix the check at the same time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20201106072551.689-2-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add delay to fix pop noise from speaker.
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20201105030804.31115-1-jack.yu@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A batch of driver specific fixes that have come up since the merge
window, nothing particularly major here but all good to have.
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Merge tag 'asoc-fix-v5.10-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.10
A batch of driver specific fixes that have come up since the merge
window, nothing particularly major here but all good to have.
The AUD2HTX is a digital module that provides a bridge between
the Audio Subsystem and the HDMI RTX Subsystem. This module
includes intermediate storage to queue SDMA transactions prior
to being synchronized and passed to the HDMI RTX Subsystem over
the Audio Link.
The AUD2HTX contains a DMA request routed to the SDMA module.
This DMA request is controlled based on the watermark level in
the 32-entry sample buffer.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1604281947-26874-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ecause clk_prepare_enable() and clk_disable_unprepare() already checked
NULL clock parameter, so the additional checks are unnecessary, just
remove them.
Signed-off-by: Xu Wang <vulab@iscas.ac.cn>
Link: https://lore.kernel.org/r/20201029090104.29552-1-vulab@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Because clk_disable_unprepare() already checked NULL clock parameter,
so the additional check is unnecessary, just remove it.
Signed-off-by: Xu Wang <vulab@iscas.ac.cn>
Link: https://lore.kernel.org/r/20201029084137.28771-1-vulab@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixes below warning when module is compiled with W=1 C=1
sound/soc/qcom/qdsp6/q6afe-clocks.c:122:18: warning: symbol 'q6afe_clks'
was not declared. Should it be static?
Fixes: 520a1c396d ("ASoC: q6afe-clocks: add q6afe clock controller")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20201105114435.22860-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
soc-pcm's dpcm_fe_dai_do_trigger() supported DRAIN commnad up to kernel
v5.4 where explicit switch(cmd) has been introduced which takes into
account all SNDRV_PCM_TRIGGER_xxx but SNDRV_PCM_TRIGGER_DRAIN. Update
switch statement to reactive support for it.
As DRAIN is somewhat unique by lacking negative/stop counterpart, bring
behaviour of dpcm_fe_dai_do_trigger() for said command back to its
pre-v5.4 state by adding it to START/RESUME/PAUSE_RELEASE group.
Fixes: acbf27746e ("ASoC: pcm: update FE/BE trigger order based on the command")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20201026100129.8216-1-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the following sparse warnings:
./mt8192-dai-i2s.c:2040:5: warning: symbol 'mt8192_dai_i2s_get_share' was not declared. Should it be static?
./mt8192-dai-i2s.c:2060:5: warning: symbol 'mt8192_dai_i2s_set_priv' was not declared. Should it be static?
./mt8192-afe-gpio.c:15:16: warning: symbol 'aud_pinctrl' was not declared. Should it be static?
./mt8192-afe-pcm.c:70:5: warning: symbol 'mt8192_get_memif_pbuf_size' was not declared. Should it be static?
./mt8192-afe-pcm.c:2137:39: warning: symbol 'mt8192_afe_component' was not declared. Should it be static?
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Zou Wei <zou_wei@huawei.com>
Acked-by: Jiaxin Yu <jiaxin.yu@mediatek.com>
Link: https://lore.kernel.org/r/1604579287-25251-1-git-send-email-zou_wei@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A build of arm64 allmodconfig with next-20201105 fails with the error:
ERROR: modpost: "mt8192_afe_gpio_request" undefined!
ERROR: modpost: "mt8192_afe_gpio_init" undefined!
Export the symbols so that mt8192-mt6359-rt1015-rt5682.ko finds it.
Signed-off-by: Sudip Mukherjee <sudipm.mukherjee@gmail.com>
Link: https://lore.kernel.org/r/20201105124747.18383-1-sudipm.mukherjee@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixes below warning when module is compiled with W=1 C=1
lpass-cpu.c:677:22: warning: symbol 'lpass_hdmi_regmap_config'
was not declared. Should it be static?
Fixes: 7cb37b7bd0 ("ASoC: qcom: Add support for lpass hdmi driver")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20201105114100.18647-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixes audio distortion on playback for the Allen&Heath
Qu-16.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201104115717.GA19046@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Almost all other allocations in ASoC API are resource managed, the only
exception is soc-topology.c.
This patchset clean ups few unnecessary functions in preparation for
change and then changes to devm_ functions for allocation.
Amadeusz Sławiński (6):
ASoC: topology: Remove unused functions from topology API
ASoC: topology: Remove multistep topology loading
ASoC: topology: Unify all device references
ASoC: topology: Change allocations to resource managed
ASoC: topology: Remove empty functions
ASoC: topology: Simplify remove_widget function
include/sound/soc-topology.h | 13 +-
sound/soc/intel/skylake/skl-topology.c | 9 +-
sound/soc/soc-topology.c | 302 +++++--------------------
sound/soc/sof/pcm.c | 2 +-
sound/soc/sof/topology.c | 4 +-
5 files changed, 57 insertions(+), 273 deletions(-)
--
2.25.1
Add TDM slot callback function to support TDM configuration
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20201104092005.2227-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that enum and mixer kcontrols are freed by resource management
framework, removing kcontrol becomes one function call, so simplify code
in remove_widget.
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20201030145427.3497990-7-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
After changing memory management to resource managed one, some of the
functions became no ops, remove them as they are no longer needed.
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20201030145427.3497990-6-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In order for topology to be resource managed, change all allocations to
be resource managed:
k*alloc -> devm_k*alloc
kstrdup -> devm_kstrdup
Exceptions where non resource managed allocation is left is
soc_tplg_dapm_widget_create(), as it uses pointer to memory locally and
frees it up after use, as well as soc_tplg_dapm_graph_elems_load(),
which has temporary pointer to table of routes.
After conversion all redundant calls in error and clean up paths were
removed.
Also removed some variables which become unneeded when there is no calls
using them.
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20201030145427.3497990-5-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In few places tplg->comp->dev is used, while everywhere else tplg->dev
is being used. Unify those uses towards tplg->dev, as it is being set to
comp->dev during initialization anyway.
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20201030145427.3497990-4-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In theory topology can be loaded in multiple steps by providing index to
snd_soc_tplg_component_load, however, from usability point of view it
doesn't make sense, as can be seen from all current users loading
topology in one go. Remove the unnecessary parameter.
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20201030145427.3497990-3-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Topology API exposes snd_soc_tplg_widget_remove and
snd_soc_tplg_widget_remove_all, but both are nowhere used. All current
users load and unload topology as a whole. As following commits
introduce resource managed memory, remove them to simplify code and
reduce maintenance burden.
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20201030145427.3497990-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Validity bits (bit 28) must not be set in order to have the samples
valid. Some controllers look for this bit and ignore the samples if it
is set.
Fixes: 06ca24e98e ("ASoC: mchp-spdiftx: add driver for S/PDIF TX Controller")
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20201104155738.68403-1-codrin.ciubotariu@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixes audio distortion on playback for the Yamaha MODX.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Tested-by: Frank Slotta <frank.slotta@posteo.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201104120705.GA19126@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Khadas audio devices ( USB_ID_VENDOR 0x3353 )
have DSD-capable implementations from XMOS
need add new usb vendor id for recognition
Signed-off-by: Artem Lapkin <art@khadas.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201103103311.5435-1-art@khadas.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS TM420 had depop circuit for headphone.
It need to turn on by COEF bit.
[ fixed the missing enum definition by tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/3d6177d7023b4783bf2793861c577ada@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This code is correct, but it should be indented one more tab.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20201103101853.GD1127762@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
davinci_mcasp_get_dt_params alway return zero, and its return value
could be ignored by the caller. So make it 'void' type to avoid the
check its return value.
Fixes: 764958f2b5 ("ASoC: ti: davinci-mcasp: Support for auxclk-fs-ratio")
Signed-off-by: Zhang Qilong <zhangqilong3@huawei.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20201102103428.32678-1-zhangqilong3@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Besides the fact that muxclk is optional, muxclk can be set using
assigned-clocks, removing the need to set it in driver. The warning is
thus unneeded, so we can transform it in a debug print, eventually to just
reflect that muxclk was not set by the driver.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20201103100554.1307190-1-codrin.ciubotariu@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This is harmless, but the "addr" comes from the user and it could lead
to a negative shift or to shift wrapping if it's too high.
Fixes: 0b00a5615d ("ALSA: hdac_ext: add hdac extended controller")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20201103101807.GC1127762@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A semicolon is not needed after a switch statement.
Signed-off-by: Tom Rix <trix@redhat.com>
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20201101171943.2305030-1-trix@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Because clk_disable_unprepare() already checked NULL clock parameter,
so the additional check is unnecessary, just remove it.
Signed-off-by: Xu Wang <vulab@iscas.ac.cn>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20201029082513.28233-1-vulab@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
First patch adds tdm 4-slot 100fs DAI setting to avoid jitter of using
64fs on CML boards. Second patch is a DMI quirk for HP Dooly.
Changes since v1:
-Add comment on Dooly's DMI quirk
Brent Lu (2):
ASoC: intel: sof_rt5682: Add support for cml_rt1015_rt5682
ASoC: intel: sof_rt5682: Add quirk for Dooly
sound/soc/intel/boards/sof_rt5682.c | 65 +++++++++++++++++--
.../intel/common/soc-acpi-intel-cml-match.c | 13 ++++
2 files changed, 73 insertions(+), 5 deletions(-)
--
2.17.1
This DMI product family string of this board is "Google_Hatch" so the
DMI quirk will take place. However, this board is using rt1015 speaker
amp instead of max98357a specified in the quirk. Therefore, we need an
new DMI quirk for this board.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20201030170559.20370-3-brent.lu@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds the driver data and updates quirk info for cml with
rt1015 speaker amp and rt5682 headset codec. Due to different mclk
frequency on JSL and CML, we need to use 4 slot TDM 100fs to avoid
the SSP m/n counter.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20201030170559.20370-2-brent.lu@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Checkpatch script produces warning:
WARNING: function definition argument 'const struct sun4i_i2s *'
should also have an identifier name.
Let's fix this by adding identifier name to get_bclk_parent_rate()
and set_fmt() callback definition.
Acked-by: Maxime Ripard <mripard@kernel.org>
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Link: https://lore.kernel.org/r/20201030144648.397824-10-peron.clem@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Because SUN4I_I2S_FIFO_CTRL_REG is volatile, writes done while the
regmap is cache-only are ignored. To work around this, move the
configuration to a callback that runs while the ASoC core has a
runtime PM reference to the device.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <mripard@kernel.org>
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Link: https://lore.kernel.org/r/20201030144648.397824-9-peron.clem@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The FIFO TX reg is volatile and sun8i i2s register
mapping is different from sun4i.
Even if in this case it's doesn't create an issue,
Avoid setting some regs that are undefined in sun8i.
Acked-by: Maxime Ripard <mripard@kernel.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Link: https://lore.kernel.org/r/20201030144648.397824-8-peron.clem@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Extend the functionality of the driver to include support of 20 and
24 bits per sample.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Acked-by: Maxime Ripard <mripard@kernel.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Link: https://lore.kernel.org/r/20201030144648.397824-7-peron.clem@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On the newer SoCs such as the H3 and A64 this is set by default
to transfer a 0 after each sample in each slot. However the A10
and A20 SoCs that this driver was developed on had a default
setting where it padded the audio gain with zeros.
This isn't a problem while we have only support for 16bit audio
but with larger sample resolution rates in the pipeline then SEXT
bits should be cleared so that they also pad at the LSB. Without
this the audio gets distorted.
Set sign extend sample for all the sunxi generations even if they
are not affected. This will keep consistency and avoid relying on
default.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <mripard@kernel.org>
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Link: https://lore.kernel.org/r/20201030144648.397824-6-peron.clem@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We are actually using a complex formula to just return a bunch of
simple values. Also this formula is wrong for sun4i when calling
get_wss() the function return 4 instead of 3.
Replace this with a simpler switch case.
Also drop the i2s params which is unused and return a simple int as
returning an error code could be out of range for an s8 and there is
no optim to return a s8 here.
Fixes: 619c15f7fa ("ASoC: sun4i-i2s: Change SR and WSS computation")
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <mripard@kernel.org>
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Link: https://lore.kernel.org/r/20201030144648.397824-5-peron.clem@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
H6 I2S is very similar to that in H3, except it supports up to 16
channels.
Signed-off-by: Jernej Skrabec <jernej.skrabec@siol.net>
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <mripard@kernel.org>
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Link: https://lore.kernel.org/r/20201030144648.397824-4-peron.clem@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As slots and slot_width can be set manually using set_tdm().
These values are then kept in sun4i_i2s struct.
So we need to check if these values are set or not.
This is not done actually and will trigger a bug.
For example, if we set to the simple soundcard in the device-tree
dai-tdm-slot-width = <32> and then start a stream using S16_LE,
currently we would calculate BCLK for 32-bit slots, but program
lrck_period for 16-bit slots, making the sample rate double what we
expected.
To fix this, we need to check if these values are set or not but as
this logic is already done by the caller. Avoid duplicating this
logic and just pass the required values as params to set_chan_cfg().
Suggested-by: Samuel Holland <samuel@sholland.org>
Acked-by: Maxime Ripard <mripard@kernel.org>
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Link: https://lore.kernel.org/r/20201030144648.397824-3-peron.clem@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Left and Right justified mode are computed using the same formula
as DSP_A and DSP_B mode.
Which is wrong and the user manual explicitly says:
LRCK_PERDIOD:
PCM Mode: Number of BCLKs within (Left + Right) channel width.
I2S/Left-Justified/Right-Justified Mode: Number of BCLKs within each
individual channel width(Left or Right)
Fix this by using the same formula as the I2S mode.
Fixes: 7ae7834ec4 ("ASoC: sun4i-i2s: Add support for DSP formats")
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Acked-by: Maxime Ripard <mripard@kernel.org>
Link: https://lore.kernel.org/r/20201030144648.397824-2-peron.clem@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Update compatible string as board compatible and device compatible
should not be same!. Make the driver inline with the new bindings.
Fixes: aa2e278554 ("ASoC: qcom: sm8250: add sound card qrb5165-rb5 support")
Reported-by: Rob Herring <robh@kernel.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20201029101550.31695-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Disable MI2S bit clock from PAUSE/STOP/SUSPEND usecase instead of
shutdown time. Acheive this by invoking clk_disable API from
cpu daiops trigger instead of cpu daiops shutdown.
Change non-atomic API "clk_prepare_enable" to atomic API
"clk_enable" in trigger, as trigger is being called from atomic context.
Fixes: 7e6799d8f8 ("ASoC: qcom: lpass-cpu: Enable MI2S BCLK and LRCLK together")
Signed-off-by: V Sujith Kumar Reddy <vsujithk@codeaurora.org>
Signed-off-by: Srinivasa Rao Mandadapu <srivasam@codeaurora.org>
Link: https://lore.kernel.org/r/1603098363-9251-1-git-send-email-srivasam@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Update SC7180 lpass_variant structure with proper I2S bitwidth
field bit positions, as bitwidth denotes 0 to 1 bits,
but previously used only 0 bit.
Signed-off-by: V Sujith Kumar Reddy <vsujithk@codeaurora.org>
Signed-off-by: Srinivasa Rao Mandadapu <srivasam@codeaurora.org>
Link: https://lore.kernel.org/r/1603798474-4897-1-git-send-email-srivasam@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The check for an error return from the call to snd_pcm_format_width
is never true as the unsigned int bitwidth can never be less than
zero. Fix this by making bitwidth an int.
Fixes: 7cb37b7bd0 ("ASoC: qcom: Add support for lpass hdmi driver")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20201028115112.109017-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
digital gain range is -84dB min to 40dB max, however this was not
correctly specified in the range.
Fix this by with correct range!
Fixes: 8c4f021d80 ("ASoC: wcd9335: add basic controls")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20201028154340.17090-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Static analysis Coverity had detected a potential array out-of-bounds
write issue due to the fact that MAX AFE port Id was set to 16 instead
of using AFE_PORT_MAX macro.
Fix this by properly using AFE_PORT_MAX macro.
Fixes: aa2e278554 ("ASoC: qcom: sm8250: add sound card qrb5165-rb5 support")
Reported-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20201028142001.22431-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
digital gain range is -84dB min to 40dB max, however this was not
correctly specified in the range.
Fix this by with correct range!
Fixes: 1cde8b8223 ("ASoC: wcd934x: add basic controls")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20201028154340.17090-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The broken jack detection should be fixed by commit a6e7d0a4bd ("ALSA:
hda: fix jack detection with Realtek codecs when in D3"), let's try
enabling runtime PM by default again.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20201027130038.16463-4-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both pm_runtime_force_suspend() and pm_runtime_force_resume() have
some implicit checks, so it can make code flow more straightforward if
we separate runtime and system suspend callbacks.
High Definition Audio Specification, 4.5.9.3 Codec Wake From System S3
states that codec can wake the system up from S3 if WAKEEN is toggled.
Since HDA controller has different wakeup settings for runtime and
system susend, we also need to explicitly disable direct-complete which
can be enabled automatically by PCI core. In addition to that, avoid
waking up codec if runtime resume is for system suspend, to not break
direct-complete for codecs.
While at it, also remove AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP, as the
original bug commit a6630529ae ("ALSA: hda: Workaround for spurious
wakeups on some Intel platforms") solves doesn't happen with this
patch.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20201027130038.16463-3-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Upon system resume, hda_codec_pm_resume() uses hda_codec_force_resume()
to resume the codec. However, pm_runtime_force_resume() won't really
resume the codec because of pm_runtime_need_not_resume() check.
Hence, hda_codec_force_resume() schedules a jackpoll work, which is to
really power up the codec.
Instead of doing that, we can use direct-complete to make the PM flow
more straightforward, and keep codec always suspended through system PM
flow if conditions are met.
On system suspend, PM core will decide what to do based on
hda_codec_pm_prepare():
- If codec is not runtime-suspended, PM core will suspend and resume the
device as normal.
- If codec is runtime-suspended, PM core will try to keep it suspended.
If it's still suspended after system resume, we use
hda_codec_pm_complete() to resume codec if it's needed.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20201027130038.16463-2-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
System boot with plugged headset. It will not detect headset Mic.
It will happen on cold boot restart resume state.
Quirk by SSID change to quirk by pin verb.
Fixes: 13468bfa8c ("ALSA: hda/realtek - set mic to auto detect on a HP AIO machine")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/f42ae1ede1cf47029ae2bef1a42caf03@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Note:
- The machine driver patch is made by the collaboration of
Cheng-Yi Chiang <cychiang@chromium.org>
Rohit kumar <rohitkr@codeaurora.org>
Ajit Pandey <ajitp@codeaurora.org>
But Ajit has left codeaurora.
Changes from v1 to v2:
- Ducumentation: Addressed all suggestions from Doug.
- Machine driver:
- Fix comment style for license.
- Sort includes.
- Remove sc7180_snd_hw_params.
- Remove sc7180_dai_init and use aux device instead for headset jack registration.
- Statically define format for Primary MI2S.
- Atomic is not a concern because there is mutex in card to make sure
startup and shutdown happen sequentially.
- Fix missing return -EINVAL in startup.
- Use static sound card.
- Use devm_kzalloc to avoid kfree.
Changes from v2 to v3:
- Ducumentation: Addressed suggestions from Srini.
- Machine driver:
- Reuse qcom_snd_parse_of to parse properties.
- Remove playback-only and capture-only.
- Misc fixes to address comments.
Changes from v3 to v4:
- Ducumentation: Addressed suggestions from Rob.
- Remove definition of dai.
- Use 'sound-dai: true' for sound-dai schema.
- Add reg property to pass 'make dt_binding_check' check although reg is not used in the driver.
- Machine driver:
- Add Reviewed-by: Tzung-Bi Shih <tzungbi@google.com>
Changes from v4 to v5:
- Documentation: Addressed suggestions from Rob.
- Add definition for "#address-cells" and "#size-cells".
- Add additionalProperties: false
- Add required properties.
Changes from v5 to v6:
- Documentation: Addressed suggestions from Rob.
- Drop contains in compatible strings.
- Only allow dai-link@[0-9]
- Remove reg ref since it has a type definition already.
Changes from v6 to v7
- Documentation:
- Add headset-jack and hdmi-jack to specify the codec
responsible for jack detection.
- HDMI codec driver:
- Use component set_jack ops instead of exporting hdmi_codec_set_jack_detect.
- Machine driver:
- Removed aux device following Stephan's suggestion.
- Use headset-jack and hdmi-jack to specify the codec
responsible for jack detection.
- Add support for HDMI(actually DP) playback.
Changes from v7 to v8
- Documentation:
- Remove headset-jack and hdmi-jack.
- Machine driver:
- Let machine driver decide whether there is a jack on the DAI.
Changes from v8 to v9
- hdmi-codec driver:
- Fixed the naming.
- Machine driver:
- Fixed unused fields.
- Moved snd_soc_card_set_drvdata
- Keep the naming of HDMI as dai name until v5 of lpass-hdmi patches.
Changes from v9 to v10
- Documentation:
- Let compatible string be more specific for board configuration to allow
for future changes.
- Machine driver:
- Fixed unused include and macro.
- Add temporary macro SC7180_LPASS_DP for future change in sc7180-lpass.h.
- Let sound card be dynamically allocated.
- Change compatible string accordingly.
Changes from v10 to v11
- Machine driver:
- Use temporary macro LPASS_DP_RX for future change in sc7180-lpass.h.
Changes from v11 to v12
- Documentation:
- Change the file and title name for new compatible string google,sc7180-trogdor.
- Change the example of model name.
- Machine driver:
- Use the definitaion of index LPASS_DP_RX in sc7180-lpass.h.
- Fix for compatible string.
- Replace a comma with semicolon.
Ajit Pandey (1):
ASoC: qcom: sc7180: Add machine driver for sound card registration
Cheng-Yi Chiang (1):
ASoC: google: dt-bindings: Add sc7180-trogdor machine bindings
.../bindings/sound/google,sc7180-trogdor.yaml | 130 +++++++++
sound/soc/qcom/Kconfig | 12 +
sound/soc/qcom/Makefile | 2 +
sound/soc/qcom/sc7180.c | 266 ++++++++++++++++++
4 files changed, 410 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml
create mode 100644 sound/soc/qcom/sc7180.c
--
2.29.0.rc2.309.g374f81d7ae-goog
Add new driver to register sound card on sc7180 trogdor board and
do the required configuration for lpass cpu dai and external codecs
connected over MI2S interfaces.
Signed-off-by: Ajit Pandey <ajitp@codeaurora.org>
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20201027032234.1705835-3-cychiang@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The Zoom UAC-2 USB audio interface provides an async playback endpoint
("1 OUT (ASYNC)") and capture endpoint ("2 IN (ASYNC)"), both with
2-channel S32_LE in 44.1, 48, 88.2, 96, 176.4, or 192
kilosamples/s. The device provides explicit feedback to adjust the
host's playback rate, but the feedback appears unstable and biased
relative to the device's capture rate.
"alsaloop -t 1000" experiences playback underruns and tries to
resample the captured audio to match the varying playback
rate. Forcing the kernel to use implicit feedback appears to
produce more stable results. This causes the host to transmit one
playback sample for each capture sample received. (Zoom North America
has been notified of this change.)
Signed-off-by: Keith Winstein <keithw@cs.stanford.edu>
Tested-by: Keith Winstein <keithw@cs.stanford.edu>
Cc: <stable@vger.kernel.org>
BugLink: https://lore.kernel.org/r/20201027071841.GA164525@trolley.csail.mit.edu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hi Mark
soc_pcm_hw_params() does rollback when failed (A),
but, it is almost same as soc_pcm_hw_free().
static int soc_pcm_hw_params(xxx)
{
...
if (ret < 0)
goto xxx_err;
...
return ret;
^ component_err:
| ...
| interface_err:
(A) ...
| codec_err:
| ...
v return ret;
}
This kind of duplicated code can be a hotbed of bugs,
thus, this patch-set share soc_pcm_hw_free() and rollback.
Kuninori Morimoto (6):
ASoC: soc.h: remove for_each_rtd_dais_rollback()
ASoC: soc-pcm: move soc_pcm_hw_free() next to soc_pcm_hw_params()
ASoC: soc-link: add mark for snd_soc_link_hw_params/free()
ASoC: soc-component: add mark for snd_soc_pcm_component_hw_params/free()
ASoC: soc-dai: add mark for snd_soc_dai_hw_params/free()
ASoC: soc-pcm: add soc_pcm_hw_clean() and call it from soc_pcm_hw_params/free()
include/sound/soc-component.h | 6 +-
include/sound/soc-dai.h | 4 +-
include/sound/soc-link.h | 3 +-
include/sound/soc.h | 7 +-
sound/soc/soc-component.c | 19 ++---
sound/soc/soc-dai.c | 13 +++-
sound/soc/soc-dapm.c | 4 +-
sound/soc/soc-link.c | 12 +++-
sound/soc/soc-pcm.c | 131 ++++++++++++++--------------------
9 files changed, 97 insertions(+), 102 deletions(-)
--
2.25.1
This series adds support the other two AIFs present in the sun8i codec,
which can be used for codec2codec DAI links.
This series first fills out the DAI driver, removing assumptions that
were made for AIF1 (16 bits, 2 channels, certain clock inversions). Some
new logic is required to handle 3 DAIs and the ADC/DAC sharing the same
clock. Finally, it adds the new DAIs, and hooks them up with DAPM
widgets and routes per the hardware topology.
To minimize the number of patches in this series, related device tree
patches (increasing #sound-dai-cells, adding new DAI links) will be sent
separately.
Changes from v1:
- Patches 1-8 from v1 (DAPM changes) were merged
- Prefixed AIF constants with "SUN8I_CODEC_" [1, 7, 10, 16, 17]
- Renamed variables in sun8i_codec_set_fmt for clarity [3]
- Update sysclk->sysclk_rate if later calls to hw_params change the
sample rate (thanks Chen-Yu for reminding me of this) [11]
- Select COMMON_CLK for clk_set_rate_exclusive [12]
- Add comments and hopefully clarify the clock protection logic [12]
- Make the error message more concise and put it on one line [12]
- Drop the "reg" variable holding SUN8I_AIF_CLK_CTRL(dai->id) [15]
- Rename "div_reg" to "clk_reg" and adjust comments for clarity [17]
- Improve the AIF2/AIF3 rate mismatch error message [17]
- Minor grammar/wording fixes in commit messages [2, 4, 7, 8, 16, 17]
- Added Maxime's Acked-by: [1-2, 4-9, 11, 13-14, 16]
Samuel Holland (17):
ASoC: sun8i-codec: Prepare to extend the DAI driver
ASoC: sun8i-codec: Program DAI format before clock inversion
ASoC: sun8i-codec: Enable all supported clock inversions
ASoC: sun8i-codec: Use the provided word size
ASoC: sun8i-codec: Round up the LRCK divisor
ASoC: sun8i-codec: Correct the BCLK divisor calculation
ASoC: sun8i-codec: Support the TDM slot binding
ASoC: sun8i-codec: Enforce symmetric DAI parameters
ASoC: sun8i-codec: Enable all supported sample rates
ASoC: sun8i-codec: Automatically set the system sample rate
ASoC: sun8i-codec: Constrain to compatible sample rates
ASoC: sun8i-codec: Protect the clock rate while streams are open
ASoC: sun8i-codec: Require an exact BCLK divisor match
ASoC: sun8i-codec: Enable all supported PCM formats
ASoC: sun8i-codec: Generalize AIF clock control
ASoC: sun8i-codec: Add the AIF2 DAI, widgets, and routes
ASoC: sun8i-codec: Add the AIF3 DAI, widgets, and routes
sound/soc/sunxi/Kconfig | 1 +
sound/soc/sunxi/sun8i-codec.c | 834 ++++++++++++++++++++++++++++------
2 files changed, 699 insertions(+), 136 deletions(-)
--
2.26.2
Add 88200Hz and 176400Hz sample rates support for TX.
Add 88200Hz, 176400Hz, 192000Hz sample rates support for RX.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/1602557360-18795-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On i.MX8QM, there are separate interrupts for TX and RX.
As the EDMA can't be configured to swing back to first FIFO
after writing the second FIFO, so we need to force the burst
size to be 2 on i.MX8QM. And EDMA don't support to shift
the data from S24_LE to S16_LE, so the supported TX format
is also different on i.MX8QM.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1602739728-4433-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP supports different audio configurations other than I2S.
This patch will fix acp driver probe failure for other audio
configurations.
Tested-by: Prike Liang <Prike.Liang@amd.com>
Reviewed-by: Prike Liang <Prike.Liang@amd.com>
Acked-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/1603476441-3506-1-git-send-email-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace the two-step copy-and-convert in
wm5102_out_comp_coeff_put() with get_unaligned_be16(). Apart from
looking nicer, it avoids this sparse warning:
wm5102.c:687:35: sparse: sparse: cast to restricted __be16
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20201015102703.24622-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The change removes the platform_data include/definition. It only contains
some values for the MICBIAS.
These are moved into 'dt-bindings/sound/adi,adau1977.h' so that they can be
used inside device-trees. When moving then, they need to be converted to
pre-compiler defines, so that the DT compiler can understand them.
The driver then, also needs to include the new
'dt-bindings/sound/adi,adau1977.h' file.
Signed-off-by: Alexandru Ardelean <alexandru.ardelean@analog.com>
Link: https://lore.kernel.org/r/20201019105313.24862-1-alexandru.ardelean@analog.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The TI PCM5102A codec driver can be used with the generic sound card
drivers, so it should be selectable. For example, with the addition
of #sound-dai-cells = <0> property in DT, it can be used with simple/graph
card drivers.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20201012141911.3150996-1-codrin.ciubotariu@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>
All platforms that use the jz4740-i2s driver have been switched to
devicetree for a while now and the assignment of the DMA peripheral ID
is done in the devicetree.
It is no longer necessary to manually assign the peripheral ID in the
driver, so remove that. The DMA driver does not even look at the value
assigned in the driver anymore and always uses the value provided by the
devicetree.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Paul Cercueil <paul@crapouillou.net>
Link: https://lore.kernel.org/r/20201023092346.5777-1-lars@metafoo.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support to Qualcomm Robotics RB5 Development Kit based on
QRB5165 Robotics SoC. This board has 2 WSA881X smart speakers
with onboard DMIC connected to internal LPASS codec via WSA
and VA macros respectively.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20201026170947.10567-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_pcm_hw_params() does rollback when failed (A),
but, it is almost same as soc_pcm_hw_free().
static int soc_pcm_hw_params(xxx)
{
...
if (ret < 0)
goto xxx_err;
...
return ret;
^ component_err:
| ...
| interface_err:
(A) ...
| codec_err:
| ...
v return ret;
}
The difference is
soc_pcm_hw_free() is for all dai/component/substream,
rollback is for succeeded part only.
This kind of duplicated code can be a hotbed of bugs,
thus, we want to share soc_pcm_hw_free() and rollback.
Now, soc_pcm_hw_params/free() are handling
1) snd_soc_link_hw_params/free()
2) snd_soc_pcm_component_hw_params/free()
3) snd_soc_dai_hw_params/free()
Now, 1) to 3) are handled.
This patch adds new soc_pcm_hw_clean() and call it from
soc_pcm_hw_params() as rollback, and from soc_pcm_hw_free() as
normal close handler.
Other difference is that soc_pcm_hw_free() handles digital mute
if it was last user. Rollback also handles it by this patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87h7rhgqab.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_pcm_hw_params() does rollback when failed (A),
but, it is almost same as soc_pcm_hw_free().
static int soc_pcm_hw_params(xxx)
{
...
if (ret < 0)
goto xxx_err;
...
return ret;
^ component_err:
| ...
| interface_err:
(A) ...
| codec_err:
| ...
v return ret;
}
The difference is
soc_pcm_hw_free() is for all dai/component/substream,
rollback is for succeeded part only.
This kind of duplicated code can be a hotbed of bugs,
thus, we want to share soc_pcm_hw_free() and rollback.
Now, soc_pcm_hw_params/free() are handling
1) snd_soc_link_hw_params/free()
2) snd_soc_pcm_component_hw_params/free()
=> 3) snd_soc_dai_hw_params/free()
This patch is for 3) snd_soc_dai_hw_params/free().
The idea of having bit-flag or counter is not enough for this purpose.
For example if one DAI is used for 2xPlaybacks for some reasons,
and if 1st Playback was succeeded but 2nd Playback was failed,
2nd Playback rollback doesn't need to call shutdown.
But it has succeeded bit-flag or counter via 1st Playback,
thus, 2nd Playback rollback will call unneeded shutdown.
And 1st Playback's necessary shutdown will not be called,
because bit-flag or counter was cleared by wrong 2nd Playback rollback.
To avoid such case, this patch marks substream pointer when hw_params() was
succeeded. If rollback needed, it will check rollback flag and marked
substream pointer.
One note here is that it cares *previous* hw_params() only now,
but we might want to check *whole* marked substream in the future.
This patch is using macro named "push/pop", so that it can be easily
update.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87imbxgqai.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_pcm_hw_params() does rollback when failed (A),
but, it is almost same as soc_pcm_hw_free().
static int soc_pcm_hw_params(xxx)
{
...
if (ret < 0)
goto xxx_err;
...
return ret;
^ component_err:
| ...
| interface_err:
(A) ...
| codec_err:
| ...
v return ret;
}
The difference is
soc_pcm_hw_free() is for all dai/component/substream,
rollback is for succeeded part only.
This kind of duplicated code can be a hotbed of bugs,
thus, we want to share soc_pcm_hw_free() and rollback.
Now, soc_pcm_hw_params/free() are handling
1) snd_soc_link_hw_params/free()
=> 2) snd_soc_pcm_component_hw_params/free()
3) snd_soc_dai_hw_params/free()
This patch is for 2) snd_soc_pcm_component_hw_params/free().
The idea of having bit-flag or counter is not enough for this purpose.
For example if one DAI is used for 2xPlaybacks for some reasons,
and if 1st Playback was succeeded but 2nd Playback was failed,
2nd Playback rollback doesn't need to call shutdown.
But it has succeeded bit-flag or counter via 1st Playback,
thus, 2nd Playback rollback will call unneeded shutdown.
And 1st Playback's necessary shutdown will not be called,
because bit-flag or counter was cleared by wrong 2nd Playback rollback.
To avoid such case, this patch marks substream pointer when hw_params() was
succeeded. If rollback needed, it will check rollback flag and marked
substream pointer.
One note here is that it cares *previous* hw_params() only now,
but we might want to check *whole* marked substream in the future.
This patch is using macro named "push/pop", so that it can be easily
update.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87k0wdgqav.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_pcm_hw_params() does rollback when failed (A),
but, it is almost same as soc_pcm_hw_free().
static int soc_pcm_hw_params(xxx)
{
...
if (ret < 0)
goto xxx_err;
...
return ret;
^ component_err:
| ...
| interface_err:
(A) ...
| codec_err:
| ...
v return ret;
}
The difference is
soc_pcm_hw_free() is for all dai/component/substream,
rollback is for succeeded part only.
This kind of duplicated code can be a hotbed of bugs,
thus, we want to share soc_pcm_hw_free() and rollback.
Now, soc_pcm_hw_params/free() are handling
=> 1) snd_soc_link_hw_params/free()
2) snd_soc_pcm_component_hw_params/free()
3) snd_soc_dai_hw_params/free()
This patch is for 1) snd_soc_link_hw_params/free().
The idea of having bit-flag or counter is not enough for this purpose.
For example if one DAI is used for 2xPlaybacks for some reasons,
and if 1st Playback was succeeded but 2nd Playback was failed,
2nd Playback rollback doesn't need to call shutdown.
But it has succeeded bit-flag or counter via 1st Playback,
thus, 2nd Playback rollback will call unneeded shutdown.
And 1st Playback's necessary shutdown will not be called,
because bit-flag or counter was cleared by wrong 2nd Playback rollback.
To avoid such case, this patch marks substream pointer when hw_params() was
succeeded. If rollback needed, it will check rollback flag and marked
substream pointer.
One note here ist that it cares *previous* hw_params() only now,
but we might want to check *whole* marked substream in the future.
This patch is using macro named "push/pop", so that it can be easily
update.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87lfgtgqba.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch moves soc_pcm_hw_free() next to soc_pcm_hw_params().
This is prepare for soc_pcm_hw_params() cleanup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87mu19gqbh.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We should only use the type, the const attribute makes no sense in
sizeof().
Reported-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Tested-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20200930152026.3902186-3-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We should only use the type, the const attribute makes no sense in
sizeof().
Reported-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Tested-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20200930152026.3902186-2-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
XCVR (Audio Transceiver) is a on-chip functional module found
on i.MX8MP. It support HDMI2.1 eARC, HDMI1.4 ARC and SPDIF.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20201013121733.83684-2-viorel.suman@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace commas with semicolons. What is done is essentially described by
the following Coccinelle semantic patch (http://coccinelle.lip6.fr/):
// <smpl>
@@ expression e1,e2; @@
e1
-,
+;
e2
... when any
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1602407979-29038-8-git-send-email-Julia.Lawall@inria.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace commas with semicolons. What is done is essentially described by
the following Coccinelle semantic patch (http://coccinelle.lip6.fr/):
// <smpl>
@@ expression e1,e2; @@
e1
-,
+;
e2
... when any
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr>
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/1602407979-29038-7-git-send-email-Julia.Lawall@inria.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace commas with semicolons. What is done is essentially described by
the following Coccinelle semantic patch (http://coccinelle.lip6.fr/):
// <smpl>
@@ expression e1,e2; @@
e1
-,
+;
e2
... when any
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr>
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/1602407979-29038-4-git-send-email-Julia.Lawall@inria.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace commas with semicolons. What is done is essentially described by
the following Coccinelle semantic patch (http://coccinelle.lip6.fr/):
// <smpl>
@@ expression e1,e2; @@
e1
-,
+;
e2
... when any
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1602407979-29038-2-git-send-email-Julia.Lawall@inria.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
AIF3 has some differences from AIF1 and AIF2:
- It supports one channel only
- It supports master mode only
- It is not directly connected to any of the mixers; instead all audio
goes through a mux with AIF2.
- It does not have its own clock dividers; instead it reuses AIF2 BCLK
and LRCK. This means that when both AIF2 and AIF3 are active, they
must use the same sample rate and total frame width. Since AIF2 and
AIF3 are only used for codec2codec DAI links, constraints are not
applicable here; the only thing we can do when the rates don't match
is report an error.
Make the necessary adjustments to support this AIF.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20201014061941.4306-18-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds support for AIF2, which is stereo and has fullly independent
clocking capability, making it very similar to AIF1.
Acked-by: Maxime Ripard <mripard@kernel.org>
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20201014061941.4306-17-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The AIF clock control register has the same layout for all three AIFs.
The only difference between them is that AIF3 is missing some fields. We
can reuse the same register field definitions for all three registers,
and use the DAI ID to select the correct register address.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Acked-by: Maxime Ripard <mripard@kernel.org>
Link: https://lore.kernel.org/r/20201014061941.4306-16-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that the DAI clock setup is correct for all hardware-supported PCM
formats, we can enable them in the driver. With the appropriate support
in the CPU DAI driver, this allows userspace to access the additional
formats.
Since this codec is connected to the CPU via a DAI, not directly, we do
not care if the CPU DAI is using 3-byte or 4-byte formats, so we can
support them both.
Acked-by: Maxime Ripard <mripard@kernel.org>
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20201014061941.4306-15-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that we guarantee that SYSCLK is running at the optimal rate when
hw_params succeeds, and that it will continue running at that rate,
SYSCLK will always be an integer multiple of BCLK. So we can always
pick the exact divider, not just the closest divider.
Acked-by: Maxime Ripard <mripard@kernel.org>
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20201014061941.4306-14-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The codec's clock input is shared among all AIFs, and shared with other
audio-related hardware in the SoC, including I2S and SPDIF controllers.
To ensure sample rates selected by userspace or by codec2codec DAI links
are maintained, the clock rate must be protected while it is in use.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Acked-by: Maxime Ripard <mripard@kernel.org>
Link: https://lore.kernel.org/r/20201014061941.4306-13-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
While another stream is active, only allow userspace to use sample rates
that are compatible with the current SYSCLK frequency. This ensures the
actual sample rate will always match what is given in hw_params.
Acked-by: Maxime Ripard <mripard@kernel.org>
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20201014061941.4306-12-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The sun8i codec has three clock/sample rate domains:
- The AIF1 domain, with a sample rate equal to AIF1 LRCK
- The AIF2 domain, with a sample rate equal to AIF2 LRCK
- The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC),
with a sample rate given by a divisor from SYSCLK. The divisor is
controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending
on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact
sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz.
When an AIF (currently only AIF1) is active, the ADC and DAC should run
at that sample rate to avoid artifacting. Sample rate conversion is only
available when multiple AIFs are active and are routed to each other;
this means the sample rate conversion hardware usually cannot be used.
Only attach the event hook to the channel 0 AIF widgets, since we only
need one event when a DAI stream starts or stops. Channel 0 is always
brought up with a DAI stream, regardless of the number of channels in
the stream.
The ADC and DAC (along with their effects blocks) can be used even if
no AIFs are in use. In that case, we should select an appropriate sample
rate divisor, instead of keeping the last-used AIF sample rate.
44.1/48 kHz was chosen to balance audio quality and power consumption.
Since the sample rate is tied to active AIF paths, disabling pmdown_time
allows switching to the optimal sample rate immediately, instead of
after a 5 second delay.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Acked-by: Maxime Ripard <mripard@kernel.org>
Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
